Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(252)

Unified Diff: webrtc/ortc/rtptransportcontroller_unittest.cc

Issue 2675173003: Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. (Closed)
Patch Set: Add memcheck suppression for end-to-end tests. Created 3 years, 10 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/ortc/rtptransportadapter.cc ('k') | webrtc/ortc/rtptransportcontrolleradapter.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/ortc/rtptransportcontroller_unittest.cc
diff --git a/webrtc/ortc/rtptransportcontroller_unittest.cc b/webrtc/ortc/rtptransportcontroller_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..40e9851d24e7e423049f7dfed7f79cb296f43909
--- /dev/null
+++ b/webrtc/ortc/rtptransportcontroller_unittest.cc
@@ -0,0 +1,195 @@
+/*
+ * Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "webrtc/base/gunit.h"
+#include "webrtc/media/base/fakemediaengine.h"
+#include "webrtc/ortc/ortcfactory.h"
+#include "webrtc/ortc/testrtpparameters.h"
+#include "webrtc/p2p/base/fakepackettransport.h"
+
+namespace webrtc {
+
+// This test uses fake packet transports and a fake media engine, in order to
+// test the RtpTransportController at only an API level. Any end-to-end test
+// should go in ortcfactory_integrationtest.cc instead.
+//
+// Currently, this test mainly focuses on the limitations of the "adapter"
+// RtpTransportController implementation. Only one of each type of
+// sender/receiver can be created, and the sender/receiver of the same media
+// type must use the same transport.
+class RtpTransportControllerTest : public testing::Test {
+ public:
+ RtpTransportControllerTest() {
+ // Note: This doesn't need to use fake network classes, since it uses
+ // FakePacketTransports.
+ auto result =
+ OrtcFactory::Create(nullptr, nullptr, nullptr, nullptr, nullptr,
+ std::unique_ptr<cricket::MediaEngineInterface>(
+ new cricket::FakeMediaEngine()));
+ ortc_factory_ = result.MoveValue();
+ rtp_transport_controller_ =
+ ortc_factory_->CreateRtpTransportController().MoveValue();
+ }
+
+ protected:
+ std::unique_ptr<OrtcFactoryInterface> ortc_factory_;
+ std::unique_ptr<RtpTransportControllerInterface> rtp_transport_controller_;
+};
+
+TEST_F(RtpTransportControllerTest, GetTransports) {
+ rtc::FakePacketTransport packet_transport1("one");
+ rtc::FakePacketTransport packet_transport2("two");
+
+ auto rtp_transport_result1 = ortc_factory_->CreateRtpTransport(
+ MakeRtcpMuxParameters(), &packet_transport1, nullptr,
+ rtp_transport_controller_.get());
+ ASSERT_TRUE(rtp_transport_result1.ok());
+
+ auto rtp_transport_result2 = ortc_factory_->CreateRtpTransport(
+ MakeRtcpMuxParameters(), &packet_transport2, nullptr,
+ rtp_transport_controller_.get());
+ ASSERT_TRUE(rtp_transport_result2.ok());
+
+ auto returned_transports = rtp_transport_controller_->GetTransports();
+ ASSERT_EQ(2u, returned_transports.size());
+ EXPECT_EQ(rtp_transport_result1.value().get(), returned_transports[0]);
+ EXPECT_EQ(rtp_transport_result2.value().get(), returned_transports[1]);
+
+ // If a transport is deleted, it shouldn't be returned any more.
+ rtp_transport_result1.MoveValue().reset();
+ returned_transports = rtp_transport_controller_->GetTransports();
+ ASSERT_EQ(1u, returned_transports.size());
+ EXPECT_EQ(rtp_transport_result2.value().get(), returned_transports[0]);
+}
+
+// Create RtpSenders and RtpReceivers on top of RtpTransports controlled by the
+// same RtpTransportController. Currently only one each of audio/video is
+// supported.
+TEST_F(RtpTransportControllerTest, AttachMultipleSendersAndReceivers) {
+ rtc::FakePacketTransport audio_packet_transport("audio");
+ rtc::FakePacketTransport video_packet_transport("video");
+
+ auto audio_rtp_transport_result = ortc_factory_->CreateRtpTransport(
+ MakeRtcpMuxParameters(), &audio_packet_transport, nullptr,
+ rtp_transport_controller_.get());
+ ASSERT_TRUE(audio_rtp_transport_result.ok());
+ auto audio_rtp_transport = audio_rtp_transport_result.MoveValue();
+
+ auto video_rtp_transport_result = ortc_factory_->CreateRtpTransport(
+ MakeRtcpMuxParameters(), &video_packet_transport, nullptr,
+ rtp_transport_controller_.get());
+ ASSERT_TRUE(video_rtp_transport_result.ok());
+ auto video_rtp_transport = video_rtp_transport_result.MoveValue();
+
+ auto audio_sender_result = ortc_factory_->CreateRtpSender(
+ cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
+ EXPECT_TRUE(audio_sender_result.ok());
+ auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
+ cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
+ EXPECT_TRUE(audio_receiver_result.ok());
+ auto video_sender_result = ortc_factory_->CreateRtpSender(
+ cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
+ EXPECT_TRUE(video_sender_result.ok());
+ auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
+ cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
+ EXPECT_TRUE(video_receiver_result.ok());
+
+ // Now that we have one each of audio/video senders/receivers, trying to
+ // create more on top of the same controller is expected to fail.
+ // TODO(deadbeef): Update this test once multiple senders/receivers on top of
+ // the same controller is supported.
+ auto failed_sender_result = ortc_factory_->CreateRtpSender(
+ cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
+ failed_sender_result.error().type());
+ auto failed_receiver_result = ortc_factory_->CreateRtpReceiver(
+ cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
+ failed_receiver_result.error().type());
+ failed_sender_result = ortc_factory_->CreateRtpSender(
+ cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
+ failed_sender_result.error().type());
+ failed_receiver_result = ortc_factory_->CreateRtpReceiver(
+ cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
+ failed_receiver_result.error().type());
+
+ // If we destroy the existing sender/receiver using a transport controller,
+ // we should be able to make a new one, despite the above limitation.
+ audio_sender_result.MoveValue().reset();
+ audio_sender_result = ortc_factory_->CreateRtpSender(
+ cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
+ EXPECT_TRUE(audio_sender_result.ok());
+ audio_receiver_result.MoveValue().reset();
+ audio_receiver_result = ortc_factory_->CreateRtpReceiver(
+ cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
+ EXPECT_TRUE(audio_receiver_result.ok());
+ video_sender_result.MoveValue().reset();
+ video_sender_result = ortc_factory_->CreateRtpSender(
+ cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
+ EXPECT_TRUE(video_sender_result.ok());
+ video_receiver_result.MoveValue().reset();
+ video_receiver_result = ortc_factory_->CreateRtpReceiver(
+ cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
+ EXPECT_TRUE(video_receiver_result.ok());
+}
+
+// Given the current limitations of the BaseChannel-based implementation, it's
+// not possible for an audio sender and receiver to use different RtpTransports.
+// TODO(deadbeef): Once this is supported, update/replace this test.
+TEST_F(RtpTransportControllerTest,
+ SenderAndReceiverUsingDifferentTransportsUnsupported) {
+ rtc::FakePacketTransport packet_transport1("one");
+ rtc::FakePacketTransport packet_transport2("two");
+
+ auto rtp_transport_result1 = ortc_factory_->CreateRtpTransport(
+ MakeRtcpMuxParameters(), &packet_transport1, nullptr,
+ rtp_transport_controller_.get());
+ ASSERT_TRUE(rtp_transport_result1.ok());
+ auto rtp_transport1 = rtp_transport_result1.MoveValue();
+
+ auto rtp_transport_result2 = ortc_factory_->CreateRtpTransport(
+ MakeRtcpMuxParameters(), &packet_transport2, nullptr,
+ rtp_transport_controller_.get());
+ ASSERT_TRUE(rtp_transport_result2.ok());
+ auto rtp_transport2 = rtp_transport_result2.MoveValue();
+
+ // Create an audio sender on transport 1, then try to create a receiver on 2.
+ auto audio_sender_result = ortc_factory_->CreateRtpSender(
+ cricket::MEDIA_TYPE_AUDIO, rtp_transport1.get());
+ EXPECT_TRUE(audio_sender_result.ok());
+ auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
+ cricket::MEDIA_TYPE_AUDIO, rtp_transport2.get());
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
+ audio_receiver_result.error().type());
+ // Delete the sender; now we should be ok to create the receiver on 2.
+ audio_sender_result.MoveValue().reset();
+ audio_receiver_result = ortc_factory_->CreateRtpReceiver(
+ cricket::MEDIA_TYPE_AUDIO, rtp_transport2.get());
+ EXPECT_TRUE(audio_receiver_result.ok());
+
+ // Do the same thing for video, reversing 1 and 2 (for variety).
+ auto video_sender_result = ortc_factory_->CreateRtpSender(
+ cricket::MEDIA_TYPE_VIDEO, rtp_transport2.get());
+ EXPECT_TRUE(video_sender_result.ok());
+ auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
+ cricket::MEDIA_TYPE_VIDEO, rtp_transport1.get());
+ EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
+ video_receiver_result.error().type());
+ video_sender_result.MoveValue().reset();
+ video_receiver_result = ortc_factory_->CreateRtpReceiver(
+ cricket::MEDIA_TYPE_VIDEO, rtp_transport1.get());
+ EXPECT_TRUE(video_receiver_result.ok());
+}
+
+} // namespace webrtc
« no previous file with comments | « webrtc/ortc/rtptransportadapter.cc ('k') | webrtc/ortc/rtptransportcontrolleradapter.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698